1922fb0ec3
In preparation for a coming refactoring CL, the (fixed) extra saturation margin is now applied into `AdaptiveModeLevelEstimator`. This CL also improves the unit tests by hard-coding its saturation params instead of reading them from a field trial. This reduces the chances of making the test flaky if a default value changes. Tested: Bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: I6765def9887a2f4e55b04d929af754cfecbb1626 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184927 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32172}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct